US5742733A - Parametric speech coding - Google Patents

Parametric speech coding Download PDF

Info

Publication number
US5742733A
US5742733A US08/382,875 US38287595A US5742733A US 5742733 A US5742733 A US 5742733A US 38287595 A US38287595 A US 38287595A US 5742733 A US5742733 A US 5742733A
Authority
US
United States
Prior art keywords
speech
signal
parameters
difference
speech signal
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Lifetime
Application number
US08/382,875
Other languages
English (en)
Inventor
Kari Juhani Jarvinen
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Qualcomm Inc
Original Assignee
Nokia Mobile Phones Ltd
Nokia Telecommunications Oy
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Nokia Mobile Phones Ltd, Nokia Telecommunications Oy filed Critical Nokia Mobile Phones Ltd
Assigned to NOKIA TELECOMMUNICATIONS OY, NOKIA MOBILE PHONES LTD. reassignment NOKIA TELECOMMUNICATIONS OY ASSIGNMENT OF ASSIGNORS INTEREST (SEE DOCUMENT FOR DETAILS). Assignors: JARVINEN, KARI JUHANI
Application granted granted Critical
Publication of US5742733A publication Critical patent/US5742733A/en
Assigned to QUALCOMM INCORPORATED reassignment QUALCOMM INCORPORATED ASSIGNMENT OF ASSIGNORS INTEREST (SEE DOCUMENT FOR DETAILS). Assignors: NOKIA CORPORATION
Assigned to NOKIA CORPORATION reassignment NOKIA CORPORATION MERGER (SEE DOCUMENT FOR DETAILS). Assignors: NOKIA MOBILE PHONES LTD.
Assigned to NOKIA NETWORKS OY reassignment NOKIA NETWORKS OY CHANGE OF NAME (SEE DOCUMENT FOR DETAILS). Assignors: NOKIA TELECOMMUNICATIONS OY
Assigned to NOKIA CORPORATION reassignment NOKIA CORPORATION MERGER (SEE DOCUMENT FOR DETAILS). Assignors: NOKIA NETWORKS OY
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Images

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters

Definitions

  • This invention relates to coding a speech signal in a coder in which a speech production model is used to calculate the excitation of the synthesis filters and the parameters of the audio channel.
  • a synthesized speech signal is generated by means of a derived excitation.
  • each phone has a speech coder/decoder (codec) which codes the speech to be transmitted and decodes the received speech.
  • codec codes the speech to be transmitted and decodes the received speech.
  • present coding methods which are combinations of waveform coding and vocoding, the compression of the signal takes place by using adaptive prediction to eliminate the short- and long-term redundance from the speech samples before quantizing the signal.
  • the coder of a GSM system is called RPE-LTP (Regular Pulse Excitation--Long Term Prediction). It uses LPC (Linear Predictive Coding) for short-term prediction and prediction of the basic frequency, that is, Long Term Prediction, LTP. The latter is used in the speech signal end also in the short-term prediction residual signal to eliminate the pronounced long-term correlation that can be perceived at the time level.
  • LTP Linear Predictive Coding
  • sampling takes place at an 8 kHz frequency and the algorithm assumes the input frame signal to be 13 bit linear PCM.
  • the samples are segmented into frames of 160 sample each frame having a duration of 20 ms.
  • the coding operations are done on a frame-specific basis or on their subframes (in blocks of 40 samples).
  • CELP Code Excited Linear Prediction
  • the actual speech signal or a residual signal filtered from it are not used as the excitation but this function is taken over by, for example, Gaussian noise, which is filtered (by shaping the spectrum) to produce speech.
  • Gaussian noise which is filtered (by shaping the spectrum) to produce speech.
  • a certain number of excitation vectors of a given length, which are comprised of random samples, are stored in the code book. These are filtered through the long- and short-term synthesis filters and the reconstructed speech signal thereby obtained is subtracted from the original speech signal.
  • the filter coefficients are obtained by analysing the original speech frame with LPC analysis and, for the LTP, by defining the basic frequency.
  • the code letter index (address) of this vector is sent together with the filter parameters to the decoder. It has the same code book as the encoder and a search is made in it, on the basis of the address, for the excitation vector indicated by the index, which excitation vector is filtered to synthesize speech in a corresponding fashion as in the encoder. No actual speech signal is thus transmitted but only filter parameters and a code book index.
  • VSELP Vector Sum Excited Linear Production
  • this method being in and of itself a method of the CELP type but which is very peculiar as to its code book. It does not permit the use as an excitation of, for example, Gaussian Noise, as in the above-described general coder of the CELP type.
  • speech coding systems are typically based on the use of a suitable speech production model.
  • the parameters according to the speech production model are calculated from the speech signal in the encoding that is to be carried out on the transmission side of a coding system of this type.
  • the values of the parameters of the speech production model are quantized and transmitted to the receiver.
  • the speech signal is synthesized using the speech production model, which is controlled with parameter values obtained from the encoder.
  • Means do not exist for fully modelling a speech signal based solely on LPC and LTP modelling, which means that in order to maintain a good quality speech signal in the coding operation, it has proved necessary to transmit to the receiver not only the parameters according to the two models mentioned but also the difference between the speech signal produced by means of the speech production model that is formed from these and the speech signal to be coded, that is, the modelling error.
  • the representation of the speech signal that is to be quantized and transmitted to the decoder is thus made up not only of a group of parameters according to the speech production model (eg, the parameters of the LPC model and the parameters of the LTP model) but also of the difference between the speech signal that is synthesized for said parameter group and the original speech signal, that is, the modelling error.
  • a parametrized representation can be formed from the modelling error or it can be quantized as such sample by sample.
  • the patent proposes the addition to the encoder of the functions of the decoder. That is to say, in accordance with the speech production model used to synthesize the speech signal, as well as of a second LPC analyser whose input is the speech signal synthesized by means of the speech production model that has been added.
  • This added LPC analyser produces other prediction parameters that describe the characteristics of the short-term spectrum of the decoded speech signal.
  • the frequency characteristics of the residual signal of the speech band are shaped according to the calculated second set of predictive parameters in such a way that a more efficient quantization is provided for the residual signal.
  • a further addition to the decoder is an LPC analyser that calculates a third set of predictive parameters which, together with the primary predictive parameters obtained from the encoder, shape the frequency characteristics of the decoded signal.
  • the arrangement eliminates the bothersome metallic background noise, or tonal noise, and enables a reduction in the bit rate.
  • the decoded speech signal obtained from a decoder according to the prior art is fed to two filters that are connected in tandem: to the first pitch filter and from there to a second adaptive spectral filter whose filter parameters are obtained from the first filter.
  • the nominator polynomial of the transfer function of the adaptive filter is proportional to the parameters of the LPC filter of the decoder and the denominator polynomial has been developed as a function of the nominator polynomial using spectral equalization technology that is known per se.
  • the denominator polynomial tracks tile nominator polynomial as well as possible, in which case the specific curve of the spectrum of the filter does not contain abnormal abrupt rises and falls that "plug up" the filter. Poor tracking causes time-dependent modulation in the decoded speech, in which case the speech is not clear.
  • a speech encoder comprising a first parametrization module for determining first prediction parameters corresponding to a speech signal input thereto, an analysis filter module for determining a modelling error corresponding to the speech signal and first prediction parameters, a synthesis filter module for forming a reconstructed speech signal corresponding to the modelling error and the first prediction parameters, a second parametrization module for determining a second set of prediction parameters corresponding to the reconstructed speech signal, a comparison module for forming a comparison signal indicative of a difference between the first and second prediction parameters, and a shaping module for shaping the modelling error such that the difference between the first and second prediction parameters is reduced, and in a second aspect there is provided a method for speech encoding comprising synthesising a second speech signal from error signals indicative of a difference between a speech signal and a first synthesised speech signal for producing a second synthesised speech signal, forming a second set of speech parameters representative of the second synthesised speech signal, comparing the second set of
  • a speech encoder comprising a first parametrization module for forming first prediction parameters representative of a speech signal, an excitation generator for forming an excitation from samples stored in a code book, synthesis filters for forming a reconstructed speech signal corresponding to the excitation and the first prediction parameters, a second parametrization module for forming a second set of prediction parameters corresponding to the reconstructed speech signal, a comparison module for forming a comparison signal indicative of a difference between the first and second prediction parameters, and a control module for forming a control signal for the excitation generator, for controlling the formation of the excitation in such a way that the first and the second prediction parameters are as close as possible to each other and in a fourth aspect there is provided a method for speech encoding, comprising; synthesising a speech signal from a code selectable from a code book having a plurality of codes and a first set of speech parameters representative of the speech signal for producing a synthesised speech signal, forming a second
  • the first prediction parameters are not transmitted to a decoder disposed in a receiver, which facilitates use by a decoder of parameter values calculated from a received speech signal, instead of the need for such parameters being transmitted from the encoder to the decoder.
  • a speech decoder comprising a synthesis filter module for forming first reconstructed speech corresponding to prediction parameters and modelling errors input to the decoder, a parametrization module for forming a second set of prediction parameters indicative of the reconstructed speech, a comparison module for forming a difference signal indicative of a difference between the first prediction parameters and the second prediction parameters, and a shaping module for processing the reconstructed speech signal, and in a sixth aspect there is provided a method for speech decoding, comprising; forming a synthesised speech signal from signals including a first set of speech parameters representative of a speech signal, defining a second set of speech parameters representative of the synthesised speech signal, comparing the first set of speech parameters with the second set of speech parameters and forming a difference signal indicative of a difference between them, and adapting the synthesised speech signal corresponding to the difference signal to reduce the difference between the first and second set of speech parameters.
  • This invention is a new parametric speech coding system in which the parametrization according to the speech production model is carried out not only for the speech signal to be coded but also for the decoded, that is, synthesized speech signal.
  • the parametric representation of the synthesized signal is compared with the parametric representation of the original speech signal and the coding functions are controlled in accordance with the difference between them.
  • the invention is applied in such a way that at first parametrization according to the speech production model used in encoding is carried out on the decoded speech signal. Next, parameter values formed from the synthesized speech signal are compared with the parameter values calculated in the encoder from the speech signal to be coded. In making the comparison some known distance measure can be used, for example, the Itakura-Saito measure between the frequency distances.
  • the coding functions are controlled by the shaping block in such a way that the difference indicated by the distance measure is made to be as small as possible.
  • an embodiment of the invention in accordance with the invention consists of three blocks: a parametrization block, a comparison block and a shaping block.
  • FIG. 1a shows an encoder of the speech coding system according to the prior art
  • FIG. 1b shows a decoder of the speech coding system according to the prior art
  • FIG. 2 is a schematic block diagram of a speech decoding system according to the invention.
  • FIG. 3 shows a speech encoding system according to the invention.
  • FIG. 4 shows a speech encoding system that operates on the analysis-synthesis principle according to the invention.
  • FIG. 1a presents an encoder (transmission side) of a known parametric speech coding system and FIG. 1b shows a decoder (receiving side).
  • the speech coding system can be a hybrid coder representing a class that is generally referred to as an RELP coder (Residual Excited Linear Prediction) in the literature.
  • RELP coder Residual Excited Linear Prediction
  • speech signal 100 that is input for coding and which is sampled, the samples being inserted in blocks, or frames, of a constant length, for example, 20 ms, undergoes a calculation of the values of the parameters of the speech production model used, this being carried out in parameter block 104. It is characteristic of parametric speech coding systems according to FIG.
  • the parameter values according to the model are quantizod in quantization block 105.
  • the quantized set of parameter values 106 that models the speech signal during each frame is transmitted to the decoder once per each frame.
  • the speech signal undergoes inverse modelling of the speech production, which serves to form, by means of the model used, the difference of the synthesized signal and the original speech signal, that is, the modelling error that has arisen in the modelling.
  • an appropriate model can be used, for example, the already mentioned LPC and LTP model.
  • the invention does not place limitations on the model to be used.
  • quantized parameter values are used in block 105 so that the effect of the quantization on the parameters of the model is also taken into account.
  • the modelling error that has resulted from use of the model must also be transmitted to the receiver.
  • the modelling error formed in block 101 is quantized in block 102 and the quantized modelling error 103 is transmitted to the decoder.
  • FIG. 1b presents the structure of the decoder of a known parametric speech coding system.
  • the parameter values 112 of the speech production model which are received via the transfer channel are supplied to speech production model 111.
  • speech production model 111 which in principle is a group of filters that synthesizes the speech signal, of which group the inverse filter is the block "inverse speech production model" of the encoder, the original speech signal 113 is formed by feeding to speech production model 111 the quantized modelling error 110 that has been received via the transfer channel.
  • FIG. 2 presents an embodiment for applying a method in accordance with the invention in a known decoder according to FIG. 1b.
  • the system in accordance with the invention can be separated out from the known speech decoder to form block 206.
  • a difference compared with the known decoding system is that in the system in accordance with the invention, parametrization is carried out on the decoded speech signal, that is, calculation of the parameter values according to the speech production model is also done on the decoded, that is, the synthesized speech signal and that the parameter values calculated from the decoded speech signal are used to shape the synthesized speech signal obtained from the speech production model.
  • the decoded speech signal that is obtained from the speech production model which is used to synthesize the speech and is known per se--this should be a speech signal similar to the original one--is brought via shaping block 202 to parametrization block 205.
  • the parametrization can be based on a known parametric model of the speech signal, for example, on LPC and LTP modelling.
  • the operation of block 205 is the same as that of block 104 in FIG. 1a, that is, both form a parametric representation from the signal brought to it for the time of each speech frame.
  • the two sets of parameters that have been calculated are compared in comparison block 204: these are the original set of parameters 203 that was calculated in the encoder and received via the transfer channel as well as the set of parameters that was calculated in parametrization block 205 and calculated from the synthesized speech signal produced by speech production model 201.
  • the result of comparing the sets of parameters that is carried out in comparison block 204 controls shaping block 202 in such a way that the objective in the shaping is to provide a shaping operation which ensures that the parameter values of the synthesized speech signal formed in the decoder and the parameter values 203 obtained from the encoder are to the largest possible extent of the same kind.
  • some known method can be used such as, for example, calculation of the Itakura-Saito distance measure, whereby the parameters are close to each other when the distance indicated by the computed distance measure is as small as possible.
  • the invention does not place any conditions on shaping block 202.
  • the operations to be carried out in it can be any suitable operations such as filtering operations, or the equivalent, that shape the envelope of the spectrum of the synthesized speech signal and its fine structure in order to minimize the distance indicated by the distance measure. Minimization of the distance measure is carried out empirically in such a way that for one decoded speech frame various shaping operations are tried out and by trial and error a search is made for a shaping operation which minimizes the distance measure used in the comparison as much as possible.
  • FIG. 3 presents an embodiment for adapting a system in accordance with the invention in the encoder.
  • the encoder can be an encoder of the RELP type and suitably may operate with the decoder in FIG. 2.
  • the encoder in FIG. 3 differs from the encoder in FIG. 1a in respect of block 310, which is shown with a dashed line.
  • parametrization block 30a a set of parameters according to a suitable speech production model is calculated from the speech signal 300 that is to be coded.
  • the speech signal is brought to inverse modelling block 301, in which the prediction error is calculated, that is, the difference between the speech signal synthesized in accordance with the model and the speech signal that is to be coded.
  • the error signal is quantized in block 302 and the quantized error signal 303 is transmitted ahead to the decoder.
  • the parameter values according to the speech production model are quantized in block 305 and the quantized parameter values are utilized in block 301.
  • the parameter values according to the speech production model are also calculated from the synthesized speech signal.
  • block 310 contains a speech production model 306, a parametrization block 307, a comparison block 308 and a shaping block 309.
  • the operation of block 310 is the following: first a reconstructed speech signal is formed again in speech production model 306 by feeding the quantized error signal 303 to the executing block (the inverse operation of block 301) of speech production model 306, in reconstructing the speech the quantized parameter values 311 are used.
  • Parametrization block 307 carries out the same operation as blocks 304, 205 and 104.
  • a comparison is made, in comparison block 308, of the parameter values calculated from the original speech signal, that is, the signal to be coded, and the parameter values calculated from the synthesized speech signal.
  • the measure describing the difference between said two calculated sets of parameter values is formed and a control signal is formed in block 301 to be supplied to block 309 that shapes the modelling error that has been formed.
  • Block 309 carries out a suitable operation, for example, filtering.
  • the operations to be carried out on the modelling error are shaped in such a way that the parameters of the speech production model (the parameters supplied by block 307), which are calculated from the synthesized speech signal, are to the greatest possible extent in accordance with the parameters calculated from the original speech signal (the parameters supplied by block 304).
  • Shaping block 309 can contain, in addition to filtering operations, operations that reduce the amount of samples to be transmitted.
  • the error signal is shaped in block 309 in such a way that by means of the quantized error signal and using speech production model 306, as much as possible of the parametric representation of the speech signal can be synthesized, which corresponds to the original speech signal, that is, the signal to be coded.
  • the operation of block 310 is carried out several times per one speech frame in such a way that in it the best possible shaping operation is sought on a trial and error basis.
  • the sample values that have been found as a result of the best shaping operation that has been found are quantized and the quantized sample values (303) are transmitted ahead to the decoder.
  • the coding to be carried out on the speech signal can best be controlled by using an embodiment of the invention in the encoder in such a way that the difference between the parametric representations calculated from the synthesized speech signal and the speech signal to be coded is very small, whereby the parameter values of the speech production model need not be quantized at all and transmitted to the decoder.
  • the speech production model to be used in the decoder parameter values calculated from the synthesized speech signal formed in the decoder can be used. In this kind of system the quantized set of parameter values 311 is not forwarded to the decoder at all.
  • FIG. 4 shows another embodiment of an encoding system in accordance with the invention.
  • FIG. 4 shows an embodiment of the invention combined with a speech coder of the analysis-synthesis type.
  • the coder can be a coder of the CELP type.
  • quantization of the modelling error signal is carried out by the so-called analysis-synthesis method in which the encoding involves seeking a quantized representation of the modelling error by synthesizing the speech signal, that is, using the speech production model.
  • any quantized representations of the modelling error can be stored, for example, in a code book. Synthesis filtering is an essential part of the encoding.
  • the operating principle in systems of this type is to make a search for the best representation of the modelling error signal in such a way that the synthesized speech signal corresponding to each possible quantized modelling error that is stored in code book 409 is formed in speech production model 404, and a difference signal between the synthesized and the original speech signal 400, which is being coded, is formed in subtraction block 403.
  • Control block 408 selects the smallest vector 401 between the signals, which has produced the difference signal and been stored in the code book, for forwarding to the decoder.
  • Parametrization of speech signal 400 that has been input for coding is carried out in block 402.
  • the set of parameters thus formed which is in accordance with the speech production model, is quantized in block 410 and the quantized parameter values are used in the speech production modelling 404.
  • the representation 401 that best resembles the signal that is to be coded and which has formed the synthesized speech signal and been stored in the code book is selected for forwarding to the receiver.
  • the synthesizing embodied in the structure of the encoder can be utilized in the manner shown in block 412, which is marked with a dashed line in FIG. 4.
  • parametrization is first carried out on the speech signal in block 407.
  • the operation of parametrization block 407 is the same as the operation of block 402 and the set of parameters formed in it in accordance with the speech production model is compared with the set of parameters formed from the speech signal to be coded in parametrization block 402.
  • the comparison is carried out by calculating the distance measure between the parametric representations of the speech production model, (eg, the Itakura-Saito measure) in comparison block 405.
  • the operation of comparison block 405 corresponds to the operation of block 308 in FIG. 3 as well as the operation of block 204 in FIG. 2.
  • the coding of the error signal is controlled by means of the control signal formed as the result of the comparison in such a way that the parameters of the speech production model calculated from the synthesized speech signal conform as much as possible to the parameters calculated from the original speech signal.
  • quantization of the error signal is carried out by synthesizing different speech signals corresponding to quantized representations of the modelling error, the difference between the model and the original speech signal, that is the error signal, is not formed at all in the encoder. For this reason a corresponding shaping operation cannot be carried out on the modelling error, as was done in the encoder in FIG. 3 by means of block 309.
  • Control of the quantization of the error signal in accordance with the invention is thus carried out according to the parametric representation of the signal to be coded and the synthesized signal by means of control block 406, which controls searches made in the code book.
  • the invention can be implemented in a number of different ways as an adjunct to known encoders and decoders, nevertheless remaining within the scope of protection defined by the accompanying claims.
  • the shaping operations to be carried out according to the control of the comparison block can be any suitable operations, as can the control method used to control the code book.
  • the quality of the speech signal produced by a coding system based on parametric speech coding can be improved first of all in the receiver by combining the system in accordance with the invention with the decoding.
  • the invention can also be applied in carrying out the encoding on the transmission side, thereby achieving a coding of the error signal that is efficient from the standpoint of the speech production model.
  • a system in accordance with the invention can be used either in the encoding to be carried out on the transmission side or in the decoding to be carried out on the receiving end or in both.
  • the quality of the speech signal produced by a speech coding system based on parametric speech coding can be improved by combining a system in accordance with the invention with the decoding.
  • an embodiment of the invention can also be applied-in carrying out the encoding, thereby achieving efficient coding of the error signal of the parametric model in general in a digital data communication system
  • a system in accordance with the invention can be used either in the encoding to be carried out on the transmission side or in the decoding to be carried out at the receiving end or in both.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Circuit For Audible Band Transducer (AREA)
US08/382,875 1994-02-08 1995-02-03 Parametric speech coding Expired - Lifetime US5742733A (en)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FI940577 1994-02-08
FI940577A FI98163C (sv) 1994-02-08 1994-02-08 Kodningssystem för parametrisk talkodning

Publications (1)

Publication Number Publication Date
US5742733A true US5742733A (en) 1998-04-21

Family

ID=8539994

Family Applications (1)

Application Number Title Priority Date Filing Date
US08/382,875 Expired - Lifetime US5742733A (en) 1994-02-08 1995-02-03 Parametric speech coding

Country Status (6)

Country Link
US (1) US5742733A (sv)
EP (1) EP0666558B1 (sv)
JP (1) JP3602593B2 (sv)
DE (1) DE69524890T2 (sv)
ES (1) ES2171175T3 (sv)
FI (1) FI98163C (sv)

Cited By (20)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5943644A (en) * 1996-06-21 1999-08-24 Ricoh Company, Ltd. Speech compression coding with discrete cosine transformation of stochastic elements
US6041298A (en) * 1996-10-09 2000-03-21 Nokia Mobile Phones, Ltd. Method for synthesizing a frame of a speech signal with a computed stochastic excitation part
US6122607A (en) * 1996-04-10 2000-09-19 Telefonaktiebolaget Lm Ericsson Method and arrangement for reconstruction of a received speech signal
US6178249B1 (en) 1998-06-18 2001-01-23 Nokia Mobile Phones Limited Attachment of a micromechanical microphone
US6199035B1 (en) 1997-05-07 2001-03-06 Nokia Mobile Phones Limited Pitch-lag estimation in speech coding
US6202045B1 (en) 1997-10-02 2001-03-13 Nokia Mobile Phones, Ltd. Speech coding with variable model order linear prediction
US6266516B1 (en) 1998-03-18 2001-07-24 Nokia Mobile Phones Limited Audio diaphragm mounting arrangements in radio telephone handsets
US6463409B1 (en) * 1998-02-23 2002-10-08 Pioneer Electronic Corporation Method of and apparatus for designing code book of linear predictive parameters, method of and apparatus for coding linear predictive parameters, and program storage device readable by the designing apparatus
US6470313B1 (en) 1998-03-09 2002-10-22 Nokia Mobile Phones Ltd. Speech coding
US6473625B1 (en) 1997-12-31 2002-10-29 Nokia Mobile Phones Limited Earpiece acoustics
US6584441B1 (en) 1998-01-21 2003-06-24 Nokia Mobile Phones Limited Adaptive postfilter
US6621910B1 (en) 1997-10-06 2003-09-16 Nokia Mobile Phones Ltd. Method and arrangement for improving leak tolerance of an earpiece in a radio device
US6707910B1 (en) 1997-09-04 2004-03-16 Nokia Mobile Phones Ltd. Detection of the speech activity of a source
US6721700B1 (en) * 1997-03-14 2004-04-13 Nokia Mobile Phones Limited Audio coding method and apparatus
US20060119589A1 (en) * 1998-06-23 2006-06-08 Immersion Corporation Haptic feedback for touchpads and other touch controls
US20070106505A1 (en) * 2003-12-01 2007-05-10 Koninkijkle Phillips Electronics N.V. Audio coding
US20100023324A1 (en) * 2008-07-10 2010-01-28 Voiceage Corporation Device and Method for Quanitizing and Inverse Quanitizing LPC Filters in a Super-Frame
US20110150229A1 (en) * 2009-06-24 2011-06-23 Arizona Board Of Regents For And On Behalf Of Arizona State University Method and system for determining an auditory pattern of an audio segment
US10431242B1 (en) * 2017-11-02 2019-10-01 Gopro, Inc. Systems and methods for identifying speech based on spectral features
US11087778B2 (en) * 2019-02-15 2021-08-10 Qualcomm Incorporated Speech-to-text conversion based on quality metric

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0878790A1 (en) * 1997-05-15 1998-11-18 Hewlett-Packard Company Voice coding system and method
DE19920501A1 (de) * 1999-05-05 2000-11-09 Nokia Mobile Phones Ltd Wiedergabeverfahren für sprachgesteuerte Systeme mit textbasierter Sprachsynthese
TWI427531B (zh) * 2010-10-05 2014-02-21 Aten Int Co Ltd 遠端管理系統及其方法

Citations (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4752956A (en) * 1984-03-07 1988-06-21 U.S. Philips Corporation Digital speech coder with baseband residual coding
US5018200A (en) * 1988-09-21 1991-05-21 Nec Corporation Communication system capable of improving a speech quality by classifying speech signals
US5115469A (en) * 1988-06-08 1992-05-19 Fujitsu Limited Speech encoding/decoding apparatus having selected encoders
US5483668A (en) * 1992-06-24 1996-01-09 Nokia Mobile Phones Ltd. Method and apparatus providing handoff of a mobile station between base stations using parallel communication links established with different time slots
US5517511A (en) * 1992-11-30 1996-05-14 Digital Voice Systems, Inc. Digital transmission of acoustic signals over a noisy communication channel
US5579433A (en) * 1992-05-11 1996-11-26 Nokia Mobile Phones, Ltd. Digital coding of speech signals using analysis filtering and synthesis filtering

Patent Citations (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4752956A (en) * 1984-03-07 1988-06-21 U.S. Philips Corporation Digital speech coder with baseband residual coding
US5115469A (en) * 1988-06-08 1992-05-19 Fujitsu Limited Speech encoding/decoding apparatus having selected encoders
US5018200A (en) * 1988-09-21 1991-05-21 Nec Corporation Communication system capable of improving a speech quality by classifying speech signals
US5579433A (en) * 1992-05-11 1996-11-26 Nokia Mobile Phones, Ltd. Digital coding of speech signals using analysis filtering and synthesis filtering
US5483668A (en) * 1992-06-24 1996-01-09 Nokia Mobile Phones Ltd. Method and apparatus providing handoff of a mobile station between base stations using parallel communication links established with different time slots
US5517511A (en) * 1992-11-30 1996-05-14 Digital Voice Systems, Inc. Digital transmission of acoustic signals over a noisy communication channel

Non-Patent Citations (6)

* Cited by examiner, † Cited by third party
Title
ICASSP 90. Tseng, "An analysis-by-synthesis linear predictive model for narrowband speech coding", pp. 209-212, vol. 1 Apr. 1990.
ICASSP 90. Tseng, An analysis by synthesis linear predictive model for narrowband speech coding , pp. 209 212, vol. 1 Apr. 1990. *
Lee, Hwang S. et al., "A Vector Quantization Adaptive Predictive Coder", IEEE Aug. 1987, vol. 3, pp. 1272-1277.
Lee, Hwang S. et al., A Vector Quantization Adaptive Predictive Coder , IEEE Aug. 1987, vol. 3, pp. 1272 1277. *
Leich, H., "Technique De Codage De La Parole", Revue HF, vol. 17, No. 1/02/03, 1 Jan. 1993, pp. 37-50.
Leich, H., Technique De Codage De La Parole , Revue HF, vol. 17, No. 1/02/03, 1 Jan. 1993, pp. 37 50. *

Cited By (27)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6122607A (en) * 1996-04-10 2000-09-19 Telefonaktiebolaget Lm Ericsson Method and arrangement for reconstruction of a received speech signal
US5943644A (en) * 1996-06-21 1999-08-24 Ricoh Company, Ltd. Speech compression coding with discrete cosine transformation of stochastic elements
US6041298A (en) * 1996-10-09 2000-03-21 Nokia Mobile Phones, Ltd. Method for synthesizing a frame of a speech signal with a computed stochastic excitation part
US7194407B2 (en) 1997-03-14 2007-03-20 Nokia Corporation Audio coding method and apparatus
US20040093208A1 (en) * 1997-03-14 2004-05-13 Lin Yin Audio coding method and apparatus
US6721700B1 (en) * 1997-03-14 2004-04-13 Nokia Mobile Phones Limited Audio coding method and apparatus
US6199035B1 (en) 1997-05-07 2001-03-06 Nokia Mobile Phones Limited Pitch-lag estimation in speech coding
US6707910B1 (en) 1997-09-04 2004-03-16 Nokia Mobile Phones Ltd. Detection of the speech activity of a source
US6202045B1 (en) 1997-10-02 2001-03-13 Nokia Mobile Phones, Ltd. Speech coding with variable model order linear prediction
US6621910B1 (en) 1997-10-06 2003-09-16 Nokia Mobile Phones Ltd. Method and arrangement for improving leak tolerance of an earpiece in a radio device
US6473625B1 (en) 1997-12-31 2002-10-29 Nokia Mobile Phones Limited Earpiece acoustics
US6584441B1 (en) 1998-01-21 2003-06-24 Nokia Mobile Phones Limited Adaptive postfilter
US6463409B1 (en) * 1998-02-23 2002-10-08 Pioneer Electronic Corporation Method of and apparatus for designing code book of linear predictive parameters, method of and apparatus for coding linear predictive parameters, and program storage device readable by the designing apparatus
US6470313B1 (en) 1998-03-09 2002-10-22 Nokia Mobile Phones Ltd. Speech coding
US6266516B1 (en) 1998-03-18 2001-07-24 Nokia Mobile Phones Limited Audio diaphragm mounting arrangements in radio telephone handsets
US6178249B1 (en) 1998-06-18 2001-01-23 Nokia Mobile Phones Limited Attachment of a micromechanical microphone
US20060119589A1 (en) * 1998-06-23 2006-06-08 Immersion Corporation Haptic feedback for touchpads and other touch controls
US20070106505A1 (en) * 2003-12-01 2007-05-10 Koninkijkle Phillips Electronics N.V. Audio coding
US20100023324A1 (en) * 2008-07-10 2010-01-28 Voiceage Corporation Device and Method for Quanitizing and Inverse Quanitizing LPC Filters in a Super-Frame
US8712764B2 (en) * 2008-07-10 2014-04-29 Voiceage Corporation Device and method for quantizing and inverse quantizing LPC filters in a super-frame
US9245532B2 (en) 2008-07-10 2016-01-26 Voiceage Corporation Variable bit rate LPC filter quantizing and inverse quantizing device and method
USRE49363E1 (en) 2008-07-10 2023-01-10 Voiceage Corporation Variable bit rate LPC filter quantizing and inverse quantizing device and method
US20110150229A1 (en) * 2009-06-24 2011-06-23 Arizona Board Of Regents For And On Behalf Of Arizona State University Method and system for determining an auditory pattern of an audio segment
US9055374B2 (en) 2009-06-24 2015-06-09 Arizona Board Of Regents For And On Behalf Of Arizona State University Method and system for determining an auditory pattern of an audio segment
US10431242B1 (en) * 2017-11-02 2019-10-01 Gopro, Inc. Systems and methods for identifying speech based on spectral features
US10546598B2 (en) * 2017-11-02 2020-01-28 Gopro, Inc. Systems and methods for identifying speech based on spectral features
US11087778B2 (en) * 2019-02-15 2021-08-10 Qualcomm Incorporated Speech-to-text conversion based on quality metric

Also Published As

Publication number Publication date
EP0666558A2 (en) 1995-08-09
JP3602593B2 (ja) 2004-12-15
EP0666558B1 (en) 2002-01-09
DE69524890T2 (de) 2003-04-10
FI98163B (sv) 1997-01-15
ES2171175T3 (es) 2002-09-01
DE69524890D1 (de) 2002-02-14
FI940577A0 (sv) 1994-02-08
FI940577A (sv) 1995-08-09
JPH0850500A (ja) 1996-02-20
EP0666558A3 (en) 1997-07-30
FI98163C (sv) 1997-04-25

Similar Documents

Publication Publication Date Title
US5742733A (en) Parametric speech coding
KR100769508B1 (ko) Ｃｅｌｐ 트랜스코딩
JP4927257B2 (ja) 可変レートスピーチ符号化
EP0409239B1 (en) Speech coding/decoding method
KR100615113B1 (ko) 주기적 음성 코딩
US7222069B2 (en) Voice code conversion apparatus
US7426465B2 (en) Speech signal decoding method and apparatus using decoded information smoothed to produce reconstructed speech signal to enhanced quality
JP4489960B2 (ja) 音声の無声セグメントの低ビットレート符号化
JPH10187197A (ja) 音声符号化方法及び該方法を実施する装置
JP4874464B2 (ja) 遷移音声フレームのマルチパルス補間的符号化
KR20010087391A (ko) 시간 동기식 파형 보간법을 이용한 피치 프로토타입파형으로부터의 음성 합성
AU669788B2 (en) Method for generating a spectral noise weighting filter for use in a speech coder
CA2293165A1 (en) Method for transmitting data in wireless speech channels
KR20040045586A (ko) 서로 다른 대역폭을 갖는 켈프 방식 코덱들 간의상호부호화 장치 및 그 방법
US7089180B2 (en) Method and device for coding speech in analysis-by-synthesis speech coders
JPH0782360B2 (ja) 音声分析合成方法
KR100341398B1 (ko) 씨이엘피형 보코더의 코드북 검색 방법
JPH09244695A (ja) 音声符号化装置及び復号化装置
Drygajilo Speech Coding Techniques and Standards
KR20050007854A (ko) 서로 다른 ｃｅｌｐ 방식의 음성 코덱 간의 상호부호화장치 및 그 방법
JP3350340B2 (ja) 音声符号化方法および音声復号化方法
KR100389898B1 (ko) 음성부호화에 있어서 선스펙트럼쌍 계수의 양자화 방법
JPH09120300A (ja) ベクトル量子化装置
JPH08160996A (ja) 音声符号化装置
EP1212750A1 (en) Multimode vselp speech coder

Legal Events

Date Code Title Description
AS Assignment

Owner name: NOKIA MOBILE PHONES LTD., FINLAND

Free format text: ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNOR:JARVINEN, KARI JUHANI;REEL/FRAME:007460/0912

Effective date: 19950328

Owner name: NOKIA TELECOMMUNICATIONS OY, FINLAND

Free format text: ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNOR:JARVINEN, KARI JUHANI;REEL/FRAME:007460/0912

Effective date: 19950328

STCF Information on status: patent grant

Free format text: PATENTED CASE

FPAY Fee payment

Year of fee payment: 4

FPAY Fee payment

Year of fee payment: 8

AS Assignment

Owner name: QUALCOMM INCORPORATED, CALIFORNIA

Free format text: ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNOR:NOKIA CORPORATION;REEL/FRAME:021998/0842

Effective date: 20081028

AS Assignment

Owner name: NOKIA CORPORATION, FINLAND

Free format text: MERGER;ASSIGNOR:NOKIA MOBILE PHONES LTD.;REEL/FRAME:022012/0882

Effective date: 20011001

AS Assignment

Owner name: NOKIA NETWORKS OY, FINLAND

Free format text: CHANGE OF NAME;ASSIGNOR:NOKIA TELECOMMUNICATIONS OY;REEL/FRAME:022343/0426

Effective date: 19991001

Owner name: NOKIA CORPORATION, FINLAND

Free format text: MERGER;ASSIGNOR:NOKIA NETWORKS OY;REEL/FRAME:022343/0431

Effective date: 20011001

FPAY Fee payment

Year of fee payment: 12